Recent Profile Visitors

Freeheel, I agree about Facebook and looking for manufacturer information there.
Facebook, for better or worse, is often the first place a customer interacts with us today. That is a challenge. Facebook is designed so users spend time on Facebook. Search is non-existent so the same questions come up every 15 days. And the FB forums these questions come up are user-generated and we have no ownership or management of them. While user-to-user data is very valuable (and there are some incredibly sharp end users) it more often than not telephone tree with incorrect information being mentioned by someone (the loudest) who presents themselves as an authoritative user.
For many years we had our own forums. Over time customers abandon it as Facebook came to be the place they spent time, since it was where all their "brands" were and didn't have to hop around.
At least for SD gear, the first place anyone should go if they have questions is to Danny, Dan, Dennis, or Sean in our tech support. Real humans with phones and email.
The others are good suggestions. I personally love FAQ's.

Borjam's comment about missing a curly bracket isn't far off...
We missed a few bugs in the 3.00 and 3.01 releases, and some of them manifest in the operation of the product. This truly guts us, and we will make this right, as we have since the early days of the first Sound Devices product with software control, the 442. Back then, the little micro-controller in the 442 controlled tone level, battery metering, limiter threshold, etc.
As John says above, it is a software-controlled world. Actually, it is far beyond software control. The latest generation of products, from us and many others, are software-defined hardware. In products like the new MixPre-3, MixPre-6 and MixPre-10T the schematic of the product is effectively built each time the unit powers up. It is truly incredible, and this is what enables these products to do so much in such a small footprint. So yes, firmware can completely control the behavior of the microphone preamp.

I'm posting this here in the equipment section. Don't flame me since this isn't an advertisement, but a notice for existing users - smiley face.
https://www.sounddevices.com/support/downloads
To apply the ambisonics plugin, which works with the MixPre-6 and MixPre-10T (only), update to 3.0 first, then download the plugin at no charge from:
https://store.sounddevices.com/product/ambisonics-plugin/

The dynamic range of a speech recordings can be quite high because of factors such as varying subject-to-microphone distance, whispers to screams, etc. You can make the case to record the complete dynamic range on both the iso track and the mix. The argument for controlling dynamic range is just as valid.
Every time a microphone is moved relative to the sound source, the amplitude changes. Great boom ops use that principle to manually control dynamic range. Mixers continually ride faders to limit dynamic range. It is all going to depend on what the scene/project/staff expect. Some post mixers like fat tracks that have little dynamics. Some like to make that decision on their own. Again, project-dependent.
As far as recording, as stated above, 24-bit file containers give us latitude to leave plenty of headroom. The analog part of a microphone preamp circuit largely determines its noise performance.

Bluetooth connection with the A10-TX only needs Bluetooth to be active on the device, no other setting are done there. When the app opens it looks for devices in proximity and they appear in the device list, which you then select. This is very similar to how Wingman connects to the 633, 664, 688, MixPre-3, MixPre-6, and MixPre-10T.

Some customers send gear in after a project for what I refer to as a "shave and haircut". This may be important if the gear was used for a seven week shoot on a windy, volcanic beach and the pots feel like crap and there is sand pouring out of the bag (happens more often than you can imagine). If the gear is well maintained and has a nice life, +1 on John and Larry's comments.

Constantin, I appreciate that you explained the test conditions in this speech test. It is important to note that, IMO, the source material doesn't come close to stressing these system's dynamic range, preamplifier noise performance, limiters (or lack of limiters?), and overall frequency response linearity. I believe I can identify each of the systems based more on a visual representation of the files, except for one that is sonically obvious.
Choosing gear is a matter of trade-offs. Using wireless versus a cable is that first trade-off. When you can, use a cable, of course. Why use a complex, expensive wireless link when a perfectly good, high quality microphone cable at $1/foot would do? Well, we have to! One could choose a system like Sennheiser's 9000 series wireless in HD uncompressed mode and not hear any appreciable sonic difference between a cable and the wireless. However, its numerous trade-offs in HD mode (64 QAM modulation, IIR) may not be worth it.
These tests are certainly very useful. At the end of the day, each of us should evaluate our tools first-hand. I encourage everyone to listen to each of these systems with their own specific setups.

I normally wouldn't jump into a thread where users are asking other users about a particular piece of gear, but since Jeff and Glenn from Zaxcom (a direct competitor) jumped here, so will I.
Based on the comment above from Glenn from Zaxcom, I believe Audio Limited (and Sound Devices) have very different philosophies when it comes to audio performance. For applications using the A10 wireless, users are not limited to solely (speech) content generated by an actor. Audio sources on the A10-TX are not limited to bandwidth-limited lavalier microphones designed for speech. We regularly see users connect high-performance condenser microphones to the A10-TX for speech and other sources, including sound effects. The A10 system operates at 44.1 kHz internally, and the system has linear response to 20 kHz.
As Glenn mentioned, fitting digital data into the limited RF spectrum is a challenge, and a balance needs to be found. The A10's 44.1 kHz sampling rate was chosen specifically because it saved a few percent of bandwidth versus a 48 kHz system. 32 kHz audio sampling (16 kHz bandwidth) was a compromise Audio Limited didn't want to make.
Remember that the microphone preamplifier and A/D converter at the wireless transmitter may well be the only preamplifier and A/D in the entire audio chain. That is a benefit of digital. Audio Limited recognized that and didn't want to compromise audio performance.

Our experience over the years supports Phil's. And from the mixer/input side, T powering can introduce other complexities, specifically if cables are suspect or the microphone input does not offer T and a 48V-to-T adapter is required. Phantom is a better electrical protocol all around.

Great information here. Things certainly are evolving quickly. The changing capabilities of post and delivery are driving many of the techniques in production, whether for spatial projects or narrative/drama.

Yeah, I imagine few producers know the nuances of multi-channel audio recording. No doubt you are going to bring your expertise to the project to approach the production. In the shoot you mention were you brought in for surround/immersive/3D audio recording specifically? What are the production deliverables in these projects? Raw A-format, B-format, 5.1? Other?